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  • Is it possible for a faulty processor to cause audio static/noise?

    - by Tom
    I have a Core 2 Extreme processor I received from a friend and have set up an XBMC box using it. However, I constantly get audio static whenever playing any music or videos. Here is a video of the sound: http://www.youtube.com/watch?v=SqKQkxYRVA4 I have tried replacing everything short of the case and the processor, including cables, audio interfaces, operating systems, ram, etc, leading me to think it might be either the case shorting out the motherboards I have tried or a faulty processor. Is it possible for a faulty processor to cause audio static/noise? Any feedback would be appreciated. Edit - Here's a list of things I have tried: Reinstalling OS Installing/upgrading/repairing PulseAudio/Alsa Installing alternate OSes, straight Ubuntu, Lubuntu, Xubuntu, Arch, Mint, Windows 7 Switching audio from the external card to internal Optical, audio out through HDMI, audio out through headphones Different ports on receiver (my main desktop sounds fine on the same sound system) Different optical cables Unplugging everything unnecessary from the motherboard (1 HD, 1 Stick of Ram, 1 Keyboard) Swapping out ram Swapping out the motherboard Replacing the Graphics Card (was replaced due to fan being noisy, not specifically for this problem) Different harddrives Swapping power supply Disabling onboard audio Switching Power Cable Plugging in through surge protector Plugging into different outlet on separate circuit

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  • [SOLVED} How do I restore my audio after uninstalling Ventrilo?

    - by Marcx
    Hi, I've a Dell studio 1555 bought on september with Windows 7 64bit Professional on it. The audio device works proprerly, while listening to audio contents (from disk or internet) When I use Ventrilo, the audio from other people sounds good and I hear their voices clearly When I use any other VOIP programs like Teamspeak 3, MSN or Skype, I hear a disturbed voice, and it's impossible to comprehend something... Anyway everything worked fine until I installed Ventrilo, but removing it didn´t solve my problem. Update: Here's a sample of how I hear others people voices.. Audio Sample After some tests, also the desktop has the same problem. (I tried TeamSpeak3) Here are some details on my laptop and desktop Laptop Dell Studio 1555 Core 2 Duo P8600 2.4Ghz 4Gb Ram Dual Channel Ati HD 4570 512Mb dedicated (up to 2048) IDT High Definition Audio Desktop Motherboard Asus P5KPL-AM Dual Core CPU E5200 2.50Ghz 2x2GB PC6400 Dual Channel Ati Radeon HD 4650 512MB VIA High Definition Audio Both computers have Windows 7 Professional 64Bit. So how do I restore my audio? SOLVED The problem was in router firmware, there was a bug that recognized VoIP traffic as a DOS attack and the router grambled every packet... I've installed the newest firmware and everything is fine :)

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  • hdmi output of windows 7 does not stay alive

    - by sjors miltenburg
    I have built a server that is connected to ONLY my tv (with hdmi). If I switch my tv to hdmi and boot my server I get the expected result (windows is displayed on my tv) and I can fully interact with it, no problem. However when "some time" passes the image on my tv dissapears and it seems the server is not outputting any hdmi signal anymore. It does not wake up on mousemove or keyboard presses. I can still interact with my server through a remote desktop session. The only way to get the hdmi output back is to reboot the server. I tried to prevent the computer from "sleeping" but this did not work. I tried upgrading from windows vista to windows 7 but this also did not work. Do you guys have any other ideas?

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  • Indentation (and wrap-indentation) of debug strings output by printf()/fprintf() in C program

    - by mbaitoff
    I'm using a debug output using printf() in my functions, but the output goes to the console starting at the 1st columns. I'd like to distinguish the nesting level of functions by indenting their output strings each time I dive into the function (it's implemented easily having a static int indentlevel; variable, which is incremented at the beginning of a function, used as a space-filler-count and decremented at the end). But the flaw is that once the output line becomes too long to be wrapped at the console edge, lines' wrapped parts start at column 1 of the console. Should I take care about this, since once the output is redirected to a file, lines are, say, one-line-length, and widths of the lines depend only on the text file viewer settings?

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  • C# : Redirect console application output : How to flush the output?

    - by user93422
    I am spawning external console application and use async output redirect. as shown in this SO post My problem is it seems that the spawned process needs to produce certain amount of output before I get the OutputDataReceived event notification. I want to receive the OutputDataReceived event as soon as possible. I have a bare-bones redirecting application, and here are some observations: 1. When I call a simple 'while(true) print("X");' console application (C#) I receive output event immediately. 2. When I call a 3d party app I am trying to wrap from the command line I see the line-by-line output. 3. When I call that 3d party app from my bare-bone wrapper (see 1) - the output comes in chunks (about one page size). What happens inside that app? FYI: The app in question is a "USBee DX Data Exctarctor (Async bus) v1.0".

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  • How to write C++ audio processing applications?

    - by cesko82
    Hi everyone, I'm an Electronics and Telecommunications student, next to my graduation. I'm gonna work on a project that involves my knowledge about DSP, music and audio in general. I allready know all the basic mathematic instruments and all the stuff I need to manage it, such as FFT, circular convolution ecc ecc. I want to learn C++ programming basically for one reason: it's very important in the professional world!!! And I think it's one of the most used to write applications working with audio, especially when it's about real time processing. Ok, after this small introduction I would like to know first, which are the most used libraries to work with audio processing in c++?? I was longer looking on the web but i couldn't find a lo of working stuff. (I work under linux with eclipse CDT enviroment). Then I would like to know if there are good sources to learn how to write some working code, such as for example how to write a simple low pass filter. Basically now i will not write real time applications, I would like to start from the processing of a WAV file, or even better an MP3 file, so basically on vectors of samples. Let's say that basically for now I would like to extract the waveform from an audio file, and save it to a thumbnail or to a PNG image. Ok, for now I think it's all I would need. Any ideas, advices, libraries, books, interesting sources about that? Thanks a lot in advance for any kind of answer. Giovanni.

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  • Looping HTML5 audio on the iPhone

    - by Peeps
    I'm trying to make a HTML5 webapp that simply plays a sound over and over and over again, on my iPhone. I don't know any Obj-C to do it natively. What I have works fine, but the sound only plays once: <!DOCTYPE html> <html> <head> <title>noisemaker!</title> <meta http-equiv="content-type" content="text/html; charset=utf-8" /> <meta name="viewport" content="maximum-scale=1, minimum-scale=1, width=device-width, user-scalable=no" /> <meta name="apple-mobile-web-app-capable" content="yes" /> </head> <body> <audio src="noise.mp3" autoplay controls loop></audio> </body> </html> Is there a way to either bypass the QuickTime audio screen and loop it in the webpage, or get the QuickTime audio screen to loop the sound?

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  • Virtual audio driver (microphone)

    - by Dalamber
    Hello guys, I want to develop a virtual microphone driver. Please, do not say anything about DirectShow - that's not "the way". I need a solution that will work with any software including Skype and MSN. And DirectShow doesn't fit these requirements. I found AVStream Filter-Centric Simulated Capture Driver (avssamp.sys) in Windows 7 WDK. What I need is an audio part of it. By default it reads avssamp.wav and plays it. But this driver is registered as WDM streaming capture device. And I want it in Audio Capture Device. There are some posts in the web but they are all the same: http://www.tech-archive.net/Archive/Development/microsoft.public.development.device.drivers/2005-05/msg00124.html http://www.winvistatips.com/problem-installing-avssamp-audio-capture-sources-category-t184898.html I think registering this filter-driver as audio capture device will make Skype recognize it as a microphone and thefore I will be able to push any PCM file as if it goes from mic. If someone already faced this problem before, please help. Thanks in advance.

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  • How do I restore my audio after uninstalling Ventrilo?

    - by Marcx
    Hi, I've a Dell studio 1555 bought on september with Windows 7 64bit Professional on it. The audio device works proprerly, while listening to audio contents (from disk or internet) When I use Ventrilo, the audio from other people sounds good and I hear their voices clearly When I use any other VOIP programs like Teamspeak 3, MSN or Skype, I hear a disturbed voice, and it's impossible to comprehend something... Anyway everything worked fine until I installed Ventrilo, but removing it didn´t solve my problem. So how do I restore my audio?

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  • How do I use different audio devices for different apps in Windows 8?

    - by Eclipse
    Besides switching the default audio device, how can I send the audio from one app (say x-box music) to one audio device, and another (say the video app) to another audio device? Edit: Looking further, I found this: http://channel9.msdn.com/Events/BUILD/BUILD2011/APP-408T At 16:16, he demonstrates exactly what I'm wanting to do, but when I go to the devices charm, I get a message: "You don't have any devices that can receive content from Music".

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  • Best way to learn iphone audio queue services, step by step tutorial

    - by optician
    Hi Everyone, I'm trying to learn how to handle audio at a fairly low level with audio queue services. I have been progrmaing in memory managed languages for quite a while, and have just completed the c programing tutorial by vtc (2007). This has left me comfortable with the understanding of pointers and memory allocation, but the apple documention still leaves me wanting for a simpler implenation and explaination. Maybe I need to learn objective c and cocoa better. I have heard that this book is good. Cocoa(R) Programming for Mac(R) OS X (3rd Edition) Could someone suggest a learning path that is going to help me get an better understanding of working with audio and an iphone. I want to be able to play mp3 files back and also alter the pitch of them as they are playing. I am prepared that I may have to temporarily convert the mp3 files into pcm files to do things like that to them. Thanks everyone.

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  • Which audio library to use?

    - by Jeb
    I want to build a .Net application for processing audio, and distribute it using ClickOnce deployment. I need access to a raw audio pipeline. Which audio library should I be using? I've heard the managed libraries for DirectSound are a dead end. I need as little as possible to be installed on the client's machine. Anything outside of the ClickOnce process isn't going to work. NAudio might be a possibility, but isn't there potentially a separate driver install? There's also SlimDX. It's a shame -- the managed DirectX libraries seem to work nicely and from what I've read, DirectX can be included in the ClickOnce install.

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  • Unexpected behavior with AudioQueueServices callback while recording audio

    - by rcw3
    I'm recording a continuous stream of data using AudioQueueServices. It is my understanding that the callback will only be called when the buffer fills with data. In practice, the first callback has a full buffer, the 2nd callback is 3/4 full, the 3rd callback is full, the 4th is 3/4 full, and so on. These buffers are 8000 packets (recording 8khz audio) - so I should be getting back 1s of audio to the callback each time. I've confirmed that my audio queue buffer size is correct (and is somewhat confirmed by the behavior). What am I doing wrong? Should I be doing something in the AudioQueueNewInput with a different RunLoop? I tried but this didn't seem to make a difference... By the way, if I run in the debugger, each callback is full with 8000 samples - making me think this is a threading / timing thing.

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  • Link to audio in XHTML/EPUB

    - by wxs
    I'm looking into synchronizing an ebook in epub format (so the content is in XHTML) to an audio file. I'm thinking of putting something along the lines of: <a class="audiolink" href="sound.ogg?t=1093"></a> into the body of the document, and then have a custom epub reader that recognizes those tags and synchronizes the audio accordingly. That does seem like a bit of a hack to me though, especially the use of a special class name. Does anyone have any pointers to how this may be done in a more standards-compliant manner (or somewhere where it has been done before)? Ebooks with audio annotation seem like an idea that may already be out there.

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  • Trying to build automatic audio-conferencing capability into a WebApp

    - by Keller
    Hey all, I'm working with a team of relatively novice programmers, and we are trying to create a site that will have audio-conferencing capabilities such that whenever someone visits the page, they will immediately have audio-conferencing capabilities with everyone else on the page (5 people max). Can anyone point us in a general direction? Should we be looking into building a custom app, leveraging audio conferencing software, or trying to mimic a webex program? Would Adobe Stratus be useful in getting this kind of functionality? Does anyone have any ideas about how we would design something like this on a macro level? Sorry for the noobish question, but any guidance would be deeply appreciated. Thanks, Keller

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  • Seeking through a streamed MP3 file with HTML5 <audio> tag

    - by Kyle Slattery
    Hopefully someone can help me out with this. I'm playing around with a node.js server that streams audio to a client, and I want to create an HTML5 player. Right now, I'm streaming the code from node using chunked encoding, and if you go directly to the URL, it works great. What I'd like to do is embed this using the HTML5 <audio> tag, like so: <audio src="http://server/stream?file=123"> where /stream is the endpoint for the node server to stream the MP3. The HTML5 player loads fine in Safari and Chrome, but it doesn't allow me to seek, and Safari even says it's a "Live Broadcast". In the headers of /stream, I include the file size and file type, and the response gets ended properly. Any thoughts on how I could get around this? I certainly could just send the whole file at once, but then the player would wait until the whole thing is downloaded--I'd rather stream it.

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  • Suggestion for creating custom sound recognition software to toggle audio

    - by Parrot owner
    I need to develop a program that toggles a particular audio track on or off when it recognizes a parrot scream or screech. The software would need to recognize a particular range of sounds and allow some variations in the range (as a parrot likely won't replicate its sreeches EXACTLY each time). Example: Bird screeches, no audio. Bird stops screeching for five seconds, audio track praising the bird plays. Regular chattering needs to be ignored completely, as it is not to be discouraged. I've heard of java libraries that have speech recognition with dictionaries built in, but the software would need to be taught the particular sounds that my particular parrot makes - not words or any random bird sound. In addition as I mentioned above, it would need to allow for slight variation in the sound, as the screech will likely never be 100% identical to the recorded version. What would be the best way to go about this/what language should I look into?

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  • Server-side Audio Editor

    - by Kristen
    I am looking for an audio editor that we can use server side (ASP + IIS) We want users to be able to upload an audio file, and then offer a 10 second teaser clip to other users for download. Ideally I would like our application to be able to specify Input and Output Filename, Start and End time (or Duration), and be able to fade-in and fade-out, and equalise the volume. Maybe some audio editors have a batch edit facility, and it would just be a question of installing on the server? All the keywords I have tried putting into Google have led me on a wild goose chase, hopefully someone can help me with suggestions. Thanks.

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  • iPhone xcode - Best way to control audio from several view controllers

    - by Are Refsdal
    Hi, I am pretty new to iPhone programming. I have a navBar with three views. I need to control audio from all of the views. I only want one audio stream to play at a time. I was thinking that it would be smart to let my AppDelegate have an instance of my audioplaying class and let the three other views use that instance to control the audio. My problem is that I don´t know how my views can use the audioplaying class in my AppDelegate. Is this the best approach and if so, how? Is there a better way?

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  • How to have game audio loop at a certain point

    - by Essential
    I have a storm in my game, and so I've made an ambient audio file which slowly grows into a storm and rain fades in, which then becomes a loopable storm audio file. Here is how I've done it: // Play intro clip and merge into main loop var introTime = stormIntro.length; AudioSource.PlayClipAtPoint( stormIntro, Vector3.zero, 0.7 ); Invoke( "StormMusic", introTime ); The way I'm currently trying to do it is get the length of the storm_intro audio clip, play the clip, and then invoke storm_loop to begin after the length of the intro has completed. This kinda works, but not really because there's occasionally a gap between the two. So how can I do it so the transition is seamless?

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  • Outputting audio stream into microphone

    - by Brap
    Hey everyone. Is there a way of outputting audio from my program and redirecting that stream to the system's microphone input 'layer'? I understand this might require some low-level calls being 'Pinvoked', but are there any articles that might help me. For example, if I was to run the output audio stream of my application into Window's Sound Recorder program, it would think that the audio is coming from a microphone and thus record that. I don't want to record a stream, just output it to the device's micrphone input. Thanks for any ideas.

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  • What tool can record multiple parallel stream to files of defined size?

    - by Hauke
    I would like to record record multiple audio web streams like this one in parallel to an mp3 or wma file for a duration of several days. I would like to be able to limit the file size or the duration stored in each file. The tool can be for any operating system. I do not need anything fancy like song recognition, metadata or silence detection. I haven't been able to find such a piece of software so far. Example: Tap channel "News" results in: News-090902-0000-0100.mp3, News-090902-0100-0200.mp3, etc... Who knows what tool can do this? It can be commercial software. Link in fulltext: 88.84.145.116:8000/listen.pls

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  • hdmi AC-3 audio broke after upgrading from 11.10 to 12.04.3

    - by Jim LastName
    I just updated my MythBuntu 11.10 to 12.04.3. Now, when I try to play 5.1 content (ripped DVD), my TV (and receiver) plays a "chattering" sound. I check my receiver and the digital dolby light isn't on--it's in PCM mode. So, either the audio is getting sent as AC-3, but the TV and receiver think it's PCM or the AC-3 audio got converted to multichannel PCM and they can't handle it. My setup: hdmi cable from htpc to TV. TV has an s/pdif output to my receiver. I know TV sends AC-3 audio out correctly because I see digital dolby light come on when I view a digital TV channel and PCM come on when I view an old analog channel. I can connect s/pdif from my htpc to my receiver and the digital dolby light comes on and it can decode the audio just fine. It's just not sending it right over hdmi. Now for some hints to the issue: I noticed in MythTV audio setup when I select alsa:hdmi.... the description only lists 2 channel PCM audio capability. speaker-test -Dhdmi:PCH -c6 errors about a bad channel count (only -c2 works). Finally, I tried vlc and it does the same chattering sound. These all make me think this isn't a MythTV issue, it's something lower than that. I think the best way to troubleshoot this is to start at the drivers and check each layer, one at a time all the way to alsa. I just don't know what the layers are and how to do it. So, I need to find some audio troubleshooting guide to assist me. Or, if one doesn't exist, I'd appreciate some steps. Thanks much, Jim

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